1. Field of the Invention
The present invention relates to a sampling phase synchronizing apparatus which performs sampling phase synchronization of a received signal waveform to be supplied to a Viterbi equalizer used as an equalizer for compensating for a distortion of a received signal from a transmission channel in a receiver of a digital communication system and, more particularly, to a sampling phase synchronizing apparatus which can reduce the bit error rate of the Viterbi equalizer, and a bidirectional maximum likelihood sequence estimation scheme used therefor.
2. Description of the Related Art
In recent years, digital mobile communication systems have been rapidly developed. Upon execution of land mobile communications, a received signal undergoes a complicated and considerable distortion due to multiplexed wave transmission interference with a transmission delay caused by physical environments around a mobile station, and high-speed movement of the mobile station. The mobile station must compensate for distortion components including noise from the received signal, on which noise is further superposed, using some signal processing scheme. Waveform equalizing techniques in the digital mobile communications are techniques for compensating for such distortions, and two major techniques are available. One technique corresponds to a decision feedback equalizer, and the other technique is a Viterbi equalizer (adaptive maximum likelihood sequence estimator). The former technique has been examined and put into practical applications in terms of easy realization upon evaluation based on the computational complexity, hardware scale, and the like. The latter technique is the best of the waveform equalizing techniques, and can be put into practical applications due to remarkable development of recent LSI micropatterning techniques and advent of a high speed digital signal processor (DSP) suitable for a digital signal process.
Upon reception of a received signal sequence, the Viterbi equalizer selects only one transmitted signal sequence, which agrees most with the received signal sequence, from all possible transmitted signal sequences to be transmitted. The Viterbi equalizer operates under the premise that the channel impulse response is known by some means. Therefore, a channel impulse response estimator is indispensable in the Viterbi equalizer. The channel impulse response intends to represent a transmission circumstance between the transmitter and receiver. More specifically, it intends to represent dispersion, along a time axis, of the information transmitted from a transmitter at a certain time in a circumstance in which a multipath transmission interference occurs. In other words, if a channel impulse response is estimated in the multipath transmission circumstance, the number of transmission channels and the transmission delay time between the transmitter and receiver can be estimated.
In general, a receiver is not phase-synchronized with a transmitter since the channel impulse response is unknown. A received signal waveform is oversampling-processed in the receiver, and thereafter, a phase synchronizing process is performed to decode information data. When a decision feedback equalizer is utilized in decoding of information data in the receiver, a decision feedback equalizer which has a fractionally spaced tap directly using an oversampling signal is effective, and is popularly used. The fractionally spaced decision feedback equalizer satisfies the sampling theorem since its tap interval is 1/N of a symbol transmission period T. Therefore, since the receiver becomes insensitive to the phase synchronization of a received signal, no special sampling phase synchronizing apparatus is necessary. In the fractionally spaced decision feedback equalizer, an adaptive algorithm process scheme need only operate, so that the output from a transversal filter in which an equalizing process is performed approximates a desired value, and the tap coefficient itself of the transversal filter adaptively corrects any sampling phase shift, equalizing time shift, and the like.
In the Viterbi equalizer, accurate simulation of the channel impulse response is most important to realize a low bit error rate. The tap interval of a transversal filter which simulates the channel impulse response in the Viterbi equalizer is normally set to be T, and the transversal filter can accurately simulate the channel response only when the amount of delay dispersion generated in a transmission channel is an integer multiple of T, thus realizing the best bit error rate performance. On the other hand, when the delay dispersion amount is not an integer multiple of T, a complicated intersymbol interference is generated in a received signal. In order to accurately describe the received signal, a transversal filter which has the number of taps corresponding to a length equal to the number of transmitted signal English letters constituting the received signal is required. The length is determined by the impulse response length of a band-pass filter which is normally effective for a baseband portion. When a fractionally spaced transversal filter which is obtained by setting the tap interval of the transversal filter for simulating the channel impulse response to be T/N is adopted, the receiver is expected to be insensitive to sample phase synchronization, but the tap length definitely increases N-fold. In any case, an increase in tap length directly leads to an increase in computational complexity and an increase in equivalent noise in the Viterbi equalizer, and the equalization performance consequently deteriorates.
As described above, when the Viterbi equalizer is used in the receiver, it is not practical to increase the tap length of the transversal filter which simulates the channel impulse response, and it is required to constitute the transversal filter by the number of taps corresponding to the minimum required interval T. For example, if the delay dispersion amount in a transmission channel is within T, the tap length is 2. However, when the tap interval is T, the Viterbi equalizer must be phase-synchronized with a received signal since it cannot satisfy the sampling theorem. Originally, unless the delay dispersion amount in a transmission channel is an integer multiple of the transmission symbol transmission period T, an optimum sampling phase synchronizing time cannot be determined, and a phase corresponding to a minimum bit error rate of the decoded result can only be an optimum sampling phase synchronizing condition. However, in general, the sampling phase for minimizing the bit error rate cannot be recognized in practice since the receiver cannot measure the bit error rate. Thus, as normal means for achieving phase synchronization with a signal obtained via a transmission channel suffering delay dispersion, some measure for synchronizing with the most strongly received arrival wave is taken. For example, by executing a correlation arithmetic operation between a training sequence (or unique word) included in a TDMA slot and a received signal, the appearance time of a peak value of a correlation value is determined to be an optimum sampling phase time. However, this scheme can merely set the start portion of a slit in an optimum sampling phase state, and cannot cope with a phase variation in the slit in a radio channel with rapidly varied characteristics.
On the other hand, another example adopts a scheme for making the receiver insensitive to sampling phase synchronization by arranging two transversal filters with a tap interval T parallel to each other, and shifting the phase of a signal sequence input to one filter from that of a signal sequence input to the other filter by T/2. However, with this scheme, a scheme for optimizing the synthesizing method of the outputs from the two parallel transversal filters is complex, and this arrangement is equivalent to a single fractionally spaced transversal filter in one view. As a result, substantially the same influence as deterioration caused by an increase in tap length appears, and this arrangement does not contribute to realization of a low bit error rate although the receiver is insensitive to sampling phase synchronization. Furthermore, since signals substantially suffering from an intersymbol interference must be processed in every transmission channel state, the convergence time of an adaptive algorithm is undesirably prolonged, and equivalent noise is generated due to the presence of extra codes, thus disturbing a stable operation.
As described above, when the delay dispersion amount of the transmission channel is not an integer multiple of the transmission symbol transmission period T, no optimum sampling phase time is present, and a sampling phase time at which the bit error rate can be minimized can only be set to be an optimum sampling phase time. In TDMA type communications in which transmission/reception is performed using slots each consisting of several symbols, the length of a training sequence is a signal sequence length less than 10% of the length of the entire slot, and a very long time period is required to obtain a bit error rate with certain reliability. Most of bit errors generated in a mobile transmission channel are not random errors determined by added noise, but are burst errors caused by multipath fading. For this reason, in some slots, the training sequence does not cause errors at all, but in some other slots, bit errors are destructively generated. In consideration of this fact, bit errors must be calculated using the training sequence in still more slots, resulting in a scheme far from a practical one. Furthermore, the bit error rate as an evaluation function can only determine errors in units of bits, but cannot be finely controlled in units of samples.
In the above-mentioned means for setting the appearance time of a peak value of a correlation value to be an optimum sampling phase time by executing the correlation arithmetic operation between a training sequence included in a TDMA slot and a received signal, when the transmission channel is, e.g., a 2-ray model with a delay dispersion amount T, the peak value of the correlation value appears at time t0 and time t0+T when viewed from the reference time of a transmitter. However, the receiver side cannot distinguish time t0 and time t0+T from each other since it cannot detect the reference time of the transmitter, and the appearance time of the peak value of the correlation value can only be set to be a sampling start time. In the case of the decision feedback equalizer, if a transversal filter relatively longer than the delay dispersion amount of a transmission channel is prepared, no problem is caused by even a sampling phase determined by this scheme. The reason for this has already been described above. However, in the case of the Viterbi equalizer, the transmission channel response must be accurately simulated. In other words, the receiver must be synchronized with the reference time of the transmitter. Thus, a case will be examined below wherein the above-mentioned sampling phase synchronizing apparatus which cannot distinguish time t0 and time t0+T is used in the Viterbi equalizer. If the receiver is synchronized with time t0, the channel impulse response is evaluated using the first tap as a reference tap corresponding to time t0. In this case, since the second tap corresponds to time t0+T, the channel impulse response can be normally evaluated, and the Viterbi equalizer performs a normal operation. On the contrary, if the receiver is synchronized with time t0+T, since the channel impulse response is evaluated using the first tap as a reference tap corresponding to time t0+T, the second tap becomes insignificant, and no tap corresponding to time t0 exists. Therefore, when the transmission channel circumstance varies, and the most strongly received signal shifts from a received signal through the delayed path (received at time t0+T) to a received signal through the direct path (received at time t0) (this shift is called a shift from a non-minimum phase mode to a minimum phase mode), the first tap gradually decreases. When the reception of the delay received signal is stopped, the evaluated channel impulse response has no information. If this phenomenon occurs, since the evaluated channel impulse response has a nature equivalent to random noise, the evaluated received signal becomes random. Since the channel impulse response is adaptively and sequentially updated using this random signal, burst errors occur, resulting in deterioration of the bit error rate.